3665 Commits (b1cf1879bc55ea11b03e5c7fc7266a2a772d00a0)

Author	SHA1	Message	Date
Sean Bright	b1cf1879bc	Don't leak references if we can't create a pseudo channel for mixing in MeetMe. ... If there was a problem allocating a pseudo channel when building our meetme, we weren't destroying our user container or destroying the mutexes that we created. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304659 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	5369a22c18	Merged revisions 303676 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011) | 20 lines Fix voicemail sequencing for file based storage. A previous change was made to account for when the number of voicemail messages exceeds the max limit to be handled properly, but it caused gaps in the messages to not be properly handled. This has now been resolved. In later non 1.4 branches, it appears that resequencing wasn't even occurring due from what appears and accidental code removal. (closes issue #18498) Reported by: JJCinAZ Patches: bug18498v2.patch uploaded by jpeeler (license 325) (closes issue #18486) Reported by: bluefox Patches: bug18486.patch uploaded by jpeeler (license 325) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	0f8989a993	Merged revisions 303546 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines Fix channel redirect out of MeetMe() and other issues with channel softhangup. Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped working properly. This issue includes a patch that resolves the issue by removing a call to ast_check_hangup() from app_meetme.c. I left that in my patch, as it doesn't need to be there. However, the rest of the patch fixes this problem with or without the change to app_meetme. The key difference between what happens before and after this patch is the effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(), ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme sees this which causes it to exit as intended. Checking ast_check_hangup() caused app_meetme to exit earlier in the process, and the target of the redirect saw the condition where ast_read() returned NULL. Removing ast_check_hangup() works around the issue in app_meetme, but doesn't solve the issue if another application did the same thing. There are also other edge cases where if an application finishes at the same time that a redirect happens, the target of the redirect will think that the channel hung up. So, I made some changes in pbx.c to resolve it at a deeper level. There are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to abort the hangup process. My patch extends this to remove the END_OF_Q frame from the channel's read queue, making the "abort hangup" more complete. This same technique was used in every place where a softhangup flag was cleared. (closes issue #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: https://reviewboard.asterisk.org/r/1082/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303548 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	01dfc39eb4	Fix changes to L() flag in Dial(). ... Tony Mountifield pointed out an error I had in my patch. I was a bit too aggressive on changing 'seconds' to 'milliseconds'. So I decided to do some additioanl testing and have no changed just the appropriate lines. One line says milliseconds, and the other says seconds. Probably should change this to be either just seconds or milliseconds, but I've spent too much time on this already :) (issue #18264) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	347d4b4bdb	Merged revisions 303007 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011) | 8 lines Add new queue strategy to preserve behavior for when queue members moved to ao2. Add queue strategy called "rrordered" to mimic old behavior from when queue members were stored in a linked list. ABE-2707 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	9aee0b5e0b	Resolve a compiler warning. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	094efca065	Option L() is milliseconds, not seconds. ... > Change the verbose output of option L() to say milliseconds and not seconds > as the value is in milliseconds. > > (closes issue #18264) > Reported by: jacco > Patches: > app_dial_patch.txt uploaded by lmadsen (license 10) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	39a167a12a	Support greetingsfolder as documented in voicemail.conf.sample. ... (closes issue #17870) Reported by: edhorton Patches: __20100816-app_voicemail-greetingsfolder-support.txt uploaded by lmadsen (license 10) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	5e7a60d065	Indicate log level argument for Log() is not optional ... (closes issue #18586) Reported by: kshumard Patches: app_verbose.c.patch uploaded by kshumard (license 92) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301176 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	15bf424dcb	Initialize useropts/adminopts in case there is no column in the realtime DB. ... (closes issue #18182) Reported by: dimas Patches: v1-18182.patch uploaded by dimas (license 88) Tested by: dimas git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	1a3a5b1694	Fix regression causing forwarding voicemails to not work with file storage. ... I had actually already fixed this in 295200 in 1.4 and thought it wasn't missing in the other branches for some reason. (closes issue #18358) Reported by: cabal95 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	5068f61abd	Merged revisions 300918 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011) | 7 lines Ensure good bye prompt in voicemail is played at the correct time. Specifically in the case of timing out but not leaving voicemail nothing should be heard. And when leaving voicemail it should be heard. ABE-2647 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@300951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	1aa72d01e0	Documentation typo ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@299864 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	41ffe44640	Merged revisions 298683 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16 Dec 2010) | 2 lines After recording only silence for a voicemail prepending, restore backup files. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@298684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	43c0271f4d	Merged revisions 298596 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010) | 7 lines Fix improper hangup when doing an attended transfer to queue. Had to indicate ringing in wait_for_answer so the attended transfer code would not try and hang up the local channel it created, which would kill the call. ABE-2624 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@298597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	07fb2027c1	Merged revisions 297689 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010) | 8 lines Don't create a Local channel if the target extension does not exist. (closes issue #18126) Reported by: junky Patches: followme.diff uploaded by junky (license 177) (partially restructured by me to avoid a possible memory leak) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@297713 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	2b8d1baccf	Merged revisions 297228 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010) | 6 lines Add "DAHDI" to a couple of app_meetme error messages. This is in response to some questions on IRC. To the user, there was nothing that made it obvious that this error had anything to do with DAHDI not being loaded. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@297229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	97e335005a	Merged revisions 296868 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010) | 4 lines Properly restore backup information file when hanging up during message prepending. ABE-2654 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@296869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	130924af07	18 characters is too short for most date/times (20 is the usual, but we add more in case of greater precision). ... (closes issue #18369) Reported by: tnakonz git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@296466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	5c520fdb04	Merged revisions 296000 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines Handle failures building translation paths more effectively. The problem scenario occurred on a heavily loaded system that was using the codec_dahdi module and exceeded the hardware transcoding capacity. The failure mode at that point was not good. The report came in to us as an Asterisk lock-up. The "core show locks" shows a ton of threads locked up (but no obvious deadlock). Upon deeper investigation, when the system is in this state, the CPU was maxed out. The CPU was being consumed by the Asterisk logger spewing messages on every audio frame for calls set up after transcoder capacity was reached. The purpose of this patch is to make Asterisk handle failures to create a translation path in a more graceful manner. If we can't translate, then the call just needs to be dropped, as it's not going to work. These are the changes: 1) In set_format() of channel.c (which is called by set_read_format() and set_write_format()), it was ignoring if ast_translator_build_path() failed and returned NULL. It now pays attention to that case and returns a result reflecting failure. With this change in place, the bridging code will immediately detect a failure and end the bridge instead of proceeding to try to bridge frames that can't be translated and making channel drivers freak out by sending them frames in a format they weren't expecting. 2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was ignored. It is now reflected in the return value of the function. This didn't turn out to have any affect on the bug, but seemed like a good change to leave in. 3) In app_dial(), when only sending a call to a single endpoint, it will attempt to do some bridging of its own of early audio. It uses make_compatible() when it's going to do this. However, it ignored failure from make compatible. So, even with the fix from #1, if there was early audio going through app_dial, there would still be a period of invalid frames passing through. After detecting failure here, Dial() exits. ABE-2658 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@296001 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	da2b2990bf	Merged revisions 295790 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. To recreate the problem: 1) Party A calls Party B 2) Invoke CLI "channel redirect" command to redirect channel call leg associated with A. 3) All associated channels are hung up. Note that if the CLI command were done on the channel call leg associated with B it works. This regression was a result of the fix for issue #16946 (https://reviewboard.asterisk.org/r/740/). The regression affects all features that use an async goto to execute the dialplan because of an external event: Channel redirect, AMI redirect, SIP REFER, and FAX detection. The struct ast_channel._softhangup code is a mess. The variable is used for several purposes that do not necessarily result in the call being hung up. I have added doxygen comments to describe how the various _softhangup bits are used. I have corrected all the places where the variable was tested in a non-bit oriented manner. The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so the soft hangup requests that do not normally result in a hangup do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171) Reported by: SantaFox (closes issue #18185) Reported by: kwemheuer (closes issue #18211) Reported by: zahir_koradia (closes issue #18230) Reported by: vmarrone (closes issue #18299) Reported by: mbrevda (closes issue #18322) Reported by: nerbos Review: https://reviewboard.asterisk.org/r/1013/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@295843 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	37e3facace	Return correct error code if lock path fails. The recent changes to open_mailbox actually caused it to be fixed, but let's be consistent. ... Reported by alecdavis in asterisk-dev. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@294910 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	9d738f023c	Merged revisions 294903 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010) | 16 lines Fix regression causing abort in voicemail after opening a mailbox with no mesgs. In order to be more safe, some error handling code was changed to respect more error conditions including the potential memory allocation failure for deleted and heard message tracking introduced in 293004. However, last_message_index returns -1 for zero messages (perhaps as expected) and was triggering the stricter error checking. Because last_message_index is only called directly in one place, just return 0 from open_mailbox (for file based storage) when no messages are detected unless a real error has occurred. (closes issue #18240) Reported by: leobrown Patches: bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585) Tested by: pabelanger ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@294904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	4a93be867e	Merged revisions 293004 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010) | 29 lines Fix inprocess_container in voicemail to correctly restrict max messages. The comparison function logic was off, so the number of sessions for a given mailbox were not being incremented properly. This problem caused the maximum number of messages per folder to not be respected when simultaneously leaving multiple voicemails just below the threshold. These problems should be fixed by the above, but just in case: Fixed resequence_mailbox to rely on the actual number of detected number of files in a directory rather than just assuming only 10 messages more than the maximum had been left. Also if more messages than the maximum are deleted they are actually removed now. The second purpose of this commit should have been separated out probably, but is related to the above. Again, if the number of messages in a given voicemail folder exceeds the maximum set limit make sure to allocate enough space for the deleted and heard index tracking array. A few random fixes: There was a forgotten decrement of the inprocess count in imap_store_file. When using IMAP storage, do not look in the directory where file based storage messages may still reside and influence the message count. Ensure to use only the first format in sendmail. ABE-2516 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@293118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	ee783956ff	Merged revisions 292411 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct 2010) | 10 lines Record priv-recordintro as sln, not gsm This removes the gsm->sln step when transcoding priv-recordintro. (closes issue #18176) Reported by: pabelanger Patches: chan_sip.diff uploaded by pabelanger (license 224) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@292412 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	34e8b90a33	Merged revisions 292223 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010) | 11 lines Fix improper operator key acceptance and clean up temp recording files. This is a fix for when pressing the operator key after recording an unavailable, busy, name, or temporary message in mailbox options. The operator key should not be accepted here, but should be allowed during the message recording. If the operator key is pressed during ensure the file is saved or deleted as apporopriate. Also, ensure removal of temporary recorded files after an early hang up or when message acceptance confirmation times out. ABE-2518 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@292226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	37fb6e394c	Fixes PickupChan() not working with full channel name. ... (closes issue #18011) Reported by: schern Patches: app_directed_pickup.c.2.patch uploaded by schern (license 995) app_directed_pickup.c.trunk.patch uploaded by schern (license 995) Tested by: schern, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@290375 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f0b5b5438b	Merged revisions 289873 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010) | 8 lines When forwarding a message, a prepend means that the filesystem will always have a better copy. (closes issue #17803) Reported by: dpetersen Patches: 20100923__issue17803.diff.txt uploaded by tilghman (license 14) Tested by: dpetersen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@289874 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	2c4db59a6b	Merged revisions 289424 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010) | 8 lines Fix a crash in app_sms. Since the data being passed to the generator callback is on the stack of the SMS() application, we must ensure that the generator is stopped before the application exits. ABE-2587 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@289425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Brett Bryant	45bda16900	Merged revisions 287758 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010) | 16 lines Fix misvalidation of meetme pins in conjunction with the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a user and admin pin setup for your conference, using the user pin would gain you admin priviledges. Also, when no user pin was set, an admin pin was, the 'a' MeetMe flag wasn't used, and the user tried to enter a conference then they were still prompted for a pin and forced to hit #. (closes issue #17908) Reported by: kuj Patches: pins_2.patch uploaded by kuj (license 1111) Tested by: kuj Review: [full review board URL with trailing slash] ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@287759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	0d71188138	Merged revisions 287386 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010) | 7 lines Blank columns should get set on reload, not ignored. (closes issue #16893) Reported by: haakon Patches: 20100818__issue16893.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@287387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	70cc29b5b9	Merged revisions 286941 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010) | 7 lines Ensure mailbox is not filled to capacity before doing message forwarding. Specifically, before prompting to record a prepended message the capacity is checked first. If the mailbox is full the extension will be reprompted. ABE-2517 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@286998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Brett Bryant	68d3fea001	Fixes a bug with MeetMe where after announcing the amount of time left in a conference, if music on hold was playing, it doesn't restart. ... (closes issue #17408) Reported by: sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by sysreq (license 1009) Tested by: sysreq git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285532 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Brett Bryant	a5c986a4e4	Merged revisions 285194 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07 Sep 2010) | 10 lines Fixes voicemail.conf issues where mailboxes with passwords that don't precede a comma would throw unnecessary error messages. (closes issue #15726) Reported by: 298 Patches: M15726.diff uploaded by junky (license 177) Tested by: junky Review: [full review board URL with trailing slash] ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285196 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	00bb0abd8e	Merged revisions 284881 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010) | 5 lines Properly detect when a sound file doesn't exist ast_fileexists returns -1 for error and 0 for a non-existant file. The existing code treated missing files as though they existed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	4e046ccf08	Don't reset queue stats on a module reload. ... (closes issue #17535) Reported by: raarts Patches: 20100819__issue17535.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284631 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	7f48d17174	Fix 3 coding errors: ... 1) After we close FD, we should not be trying to write to it. 2) Call _exit(0), not exit(0), to avoid running shutdown routines in a child. 3) Use endian, not processor, detection to ensure bytes are written in the correct order. (closes issue #15706) Reported by: modelnine Patches: asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine (license 865) Tested by: gmartinez git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	563b986aef	Only set status TIMEOUT, if we have no digits. ... (closes issue #15188) Reported by: jcovert Patches: app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license 551) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281722 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	08bae3a8f3	Merged revisions 281566 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010) | 8 lines Reset visible indication after answer. (closes issue #17641) Reported by: klaus3000 Patches: ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by klaus3000 (license 65) Tested by: schmidts ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	81fe0bf8b5	Allow the pipe, but also allow the comma ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jean Galarneau	a7e01edae4	Merged revisions 280341 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) | 2 lines Fix a dsp structure leak occuring when a local channel is put into a meetme conference, then masquaraded away. ABE-2422 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280345 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	950cbd80a5	Plug a reference leak in app_queue when adding members dynamically. ... (closes issue #17738) Reported by: bobwienholt Patches: issue17738.patch uploaded by bobwienholt (license 950) Tested by: bobwienholt, seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	33637672d0	Merged revisions 279206 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) | 7 lines SIP promiscuous redirect could fail to dial the redirect. The ast_channel was created with one variable to ast_request() but the call to ast_call() that initiates the outgoing call was using a different variable. The two variables are not equivalent if the call_forward string included a channel technology specifier. e.g., SIP/200 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279207 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	891d56d31c	Merged revisions 278463 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r278463 | tilghman | 2010-07-21 10:56:05 -0500 (Wed, 21 Jul 2010) | 11 lines Ensure realtime conferences are treated the same as static conferences when trying to find an empty one. Also, parse the useropts properly, when retrieving from realtime, and add them to the existing flags. (closes issue #17502) Reported by: kenji Patches: 20100720__issue17502.diff.txt uploaded by tilghman (license 14) Tested by: kenji ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@278464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	1fc8b2f637	Merged revisions 278275 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r278275 | tilghman | 2010-07-20 17:40:19 -0500 (Tue, 20 Jul 2010) | 14 lines Merged revisions 278261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20 Jul 2010) | 7 lines Delete IMAP messages in reverse order, to ensure reordering after each expunge does not cause deletion of the wrong message. (closes issue #16350) Reported by: noahisaac Patches: 20100623__issue16350.diff.txt uploaded by tilghman (license 14) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@278276 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	481f5eddb5	Merged revisions 277488 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r277488 | jpeeler | 2010-07-16 16:16:08 -0500 (Fri, 16 Jul 2010) | 10 lines Fix reporting estimated queue hold time. Just say the number of seconds (after minutes) rather than doing some incorrect calculation with respect to minutes. (closes issue #17498) Reported by: corruptor Patches: holdesecs_bug.diff uploaded by corruptor (license 253) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@277489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	a5120a3bc0	Merged revisions 277366 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r277366 | jpeeler | 2010-07-16 14:22:49 -0500 (Fri, 16 Jul 2010) | 7 lines Add missing handling for ringing state for use with queue empty options. (closes issue #17471) Reported by: jazzy Patches: app_queue.c.diff uploaded by jazzy (license 1056) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@277377 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	1d142f35da	Merged revisions 277183 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r277183 | pabelanger | 2010-07-16 13:13:46 -0400 (Fri, 16 Jul 2010) | 15 lines Merged revisions 277182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul 2010) | 8 lines Total analysis time error with SIP and silence suppression When using app_amd with SIP providers that have silence suppression on, the iTotalTime count increases exponentially. (closes issue #17656) Reported by: juls ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@277188 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	47d6d377ae	Merged revisions 276074 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r276074 | jpeeler | 2010-07-13 12:37:40 -0500 (Tue, 13 Jul 2010) | 19 lines Merged revisions 275773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010) | 12 lines Make user removals and traversals thread safe in meetme. Race conditions present in meetme involving the user list where a lack of locking has the potential for a user to be removed during a traversal or as in the case of the reporter after checking if the list is empty could cause a crash. Fixing this was done by convering the userlist to an ao2 container. (closes issue #17390) Reported by: Vince Review: https://reviewboard.asterisk.org/r/746/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@276121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	e05044b047	Merged revisions 275028 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r275028 | mnicholson | 2010-07-09 11:05:58 -0500 (Fri, 09 Jul 2010) | 15 lines Merged revisions 275027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul 2010) | 8 lines Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx via the G option in app_dial (closes issue #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff uploaded by mnicholson (license 96) Tested by: jamicque, mnicholson ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@275029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago